This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2030 lines
68 KiB
C
2030 lines
68 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2008, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Pluggable RTP Architecture
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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/*** DOCUMENTATION
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<managerEvent language="en_US" name="RTCPSent">
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<managerEventInstance class="EVENT_FLAG_REPORTING">
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<synopsis>Raised when an RTCP packet is sent.</synopsis>
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<syntax>
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<channel_snapshot/>
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<parameter name="SSRC">
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<para>The SSRC identifier for our stream</para>
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</parameter>
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<parameter name="PT">
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<para>The type of packet for this RTCP report.</para>
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<enumlist>
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<enum name="200(SR)"/>
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<enum name="201(RR)"/>
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</enumlist>
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</parameter>
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<parameter name="To">
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<para>The address the report is sent to.</para>
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</parameter>
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<parameter name="ReportCount">
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<para>The number of reports that were sent.</para>
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<para>The report count determines the number of ReportX headers in
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the message. The X for each set of report headers will range from 0 to
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<literal>ReportCount - 1</literal>.</para>
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</parameter>
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<parameter name="SentNTP" required="false">
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<para>The time the sender generated the report. Only valid when
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PT is <literal>200(SR)</literal>.</para>
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</parameter>
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<parameter name="SentRTP" required="false">
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<para>The sender's last RTP timestamp. Only valid when PT is
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<literal>200(SR)</literal>.</para>
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</parameter>
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<parameter name="SentPackets" required="false">
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<para>The number of packets the sender has sent. Only valid when PT
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is <literal>200(SR)</literal>.</para>
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</parameter>
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<parameter name="SentOctets" required="false">
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<para>The number of bytes the sender has sent. Only valid when PT is
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<literal>200(SR)</literal>.</para>
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</parameter>
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<parameter name="ReportXSourceSSRC">
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<para>The SSRC for the source of this report block.</para>
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</parameter>
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<parameter name="ReportXFractionLost">
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<para>The fraction of RTP data packets from <literal>ReportXSourceSSRC</literal>
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lost since the previous SR or RR report was sent.</para>
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</parameter>
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<parameter name="ReportXCumulativeLost">
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<para>The total number of RTP data packets from <literal>ReportXSourceSSRC</literal>
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lost since the beginning of reception.</para>
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</parameter>
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<parameter name="ReportXHighestSequence">
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<para>The highest sequence number received in an RTP data packet from
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<literal>ReportXSourceSSRC</literal>.</para>
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</parameter>
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<parameter name="ReportXSequenceNumberCycles">
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<para>The number of sequence number cycles seen for the RTP data
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received from <literal>ReportXSourceSSRC</literal>.</para>
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</parameter>
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<parameter name="ReportXIAJitter">
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<para>An estimate of the statistical variance of the RTP data packet
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interarrival time, measured in timestamp units.</para>
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</parameter>
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<parameter name="ReportXLSR">
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<para>The last SR timestamp received from <literal>ReportXSourceSSRC</literal>.
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If no SR has been received from <literal>ReportXSourceSSRC</literal>,
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then 0.</para>
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</parameter>
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<parameter name="ReportXDLSR">
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<para>The delay, expressed in units of 1/65536 seconds, between
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receiving the last SR packet from <literal>ReportXSourceSSRC</literal>
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and sending this report.</para>
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</parameter>
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</syntax>
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</managerEventInstance>
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</managerEvent>
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<managerEvent language="en_US" name="RTCPReceived">
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<managerEventInstance class="EVENT_FLAG_REPORTING">
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<synopsis>Raised when an RTCP packet is received.</synopsis>
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<syntax>
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<channel_snapshot/>
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<parameter name="SSRC">
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<para>The SSRC identifier for the remote system</para>
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</parameter>
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='PT'])" />
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<parameter name="From">
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<para>The address the report was received from.</para>
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</parameter>
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<parameter name="RTT">
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<para>Calculated Round-Trip Time in seconds</para>
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</parameter>
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<parameter name="ReportCount">
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<para>The number of reports that were received.</para>
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<para>The report count determines the number of ReportX headers in
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the message. The X for each set of report headers will range from 0 to
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<literal>ReportCount - 1</literal>.</para>
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</parameter>
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentNTP'])" />
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentRTP'])" />
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentPackets'])" />
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentOctets'])" />
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<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[contains(@name, 'ReportX')])" />
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</syntax>
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</managerEventInstance>
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</managerEvent>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <math.h>
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#include "asterisk/channel.h"
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#include "asterisk/frame.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/manager.h"
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#include "asterisk/options.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/pbx.h"
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#include "asterisk/translate.h"
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#include "asterisk/netsock2.h"
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#include "asterisk/_private.h"
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#include "asterisk/framehook.h"
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#include "asterisk/stasis.h"
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#include "asterisk/json.h"
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#include "asterisk/stasis_channels.h"
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struct ast_srtp_res *res_srtp = NULL;
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struct ast_srtp_policy_res *res_srtp_policy = NULL;
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/*! Structure that represents an RTP session (instance) */
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struct ast_rtp_instance {
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/*! Engine that is handling this RTP instance */
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struct ast_rtp_engine *engine;
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/*! Data unique to the RTP engine */
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void *data;
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/*! RTP properties that have been set and their value */
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int properties[AST_RTP_PROPERTY_MAX];
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/*! Address that we are expecting RTP to come in to */
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struct ast_sockaddr local_address;
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/*! Address that we are sending RTP to */
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struct ast_sockaddr remote_address;
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/*! Instance that we are bridged to if doing remote or local bridging */
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struct ast_rtp_instance *bridged;
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/*! Payload and packetization information */
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struct ast_rtp_codecs codecs;
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/*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
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int timeout;
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/*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
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int holdtimeout;
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/*! RTP keepalive interval */
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int keepalive;
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/*! Glue currently in use */
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struct ast_rtp_glue *glue;
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/*! SRTP info associated with the instance */
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struct ast_srtp *srtp;
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/*! Channel unique ID */
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char channel_uniqueid[AST_MAX_UNIQUEID];
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};
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/*! List of RTP engines that are currently registered */
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static AST_RWLIST_HEAD_STATIC(engines, ast_rtp_engine);
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/*! List of RTP glues */
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static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
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/*! The following array defines the MIME Media type (and subtype) for each
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of our codecs, or RTP-specific data type. */
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static struct ast_rtp_mime_type {
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struct ast_rtp_payload_type payload_type;
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char *type;
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char *subtype;
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unsigned int sample_rate;
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} ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
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static ast_rwlock_t mime_types_lock;
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static int mime_types_len = 0;
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/*!
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* \brief Mapping between Asterisk codecs and rtp payload types
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*
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* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
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* also, our own choices for dynamic payload types. This is our master
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* table for transmission
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*
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* See http://www.iana.org/assignments/rtp-parameters for a list of
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* assigned values
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*/
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static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
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static ast_rwlock_t static_RTP_PT_lock;
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/*! \brief \ref stasis topic for RTP related messages */
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static struct stasis_topic *rtp_topic;
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int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
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{
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struct ast_rtp_engine *current_engine;
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/* Perform a sanity check on the engine structure to make sure it has the basics */
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if (ast_strlen_zero(engine->name) || !engine->new || !engine->destroy || !engine->write || !engine->read) {
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ast_log(LOG_WARNING, "RTP Engine '%s' failed sanity check so it was not registered.\n", !ast_strlen_zero(engine->name) ? engine->name : "Unknown");
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return -1;
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}
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/* Link owner module to the RTP engine for reference counting purposes */
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engine->mod = module;
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AST_RWLIST_WRLOCK(&engines);
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/* Ensure that no two modules with the same name are registered at the same time */
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AST_RWLIST_TRAVERSE(&engines, current_engine, entry) {
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if (!strcmp(current_engine->name, engine->name)) {
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ast_log(LOG_WARNING, "An RTP engine with the name '%s' has already been registered.\n", engine->name);
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AST_RWLIST_UNLOCK(&engines);
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return -1;
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}
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}
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/* The engine survived our critique. Off to the list it goes to be used */
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AST_RWLIST_INSERT_TAIL(&engines, engine, entry);
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AST_RWLIST_UNLOCK(&engines);
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ast_verb(2, "Registered RTP engine '%s'\n", engine->name);
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return 0;
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}
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int ast_rtp_engine_unregister(struct ast_rtp_engine *engine)
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{
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struct ast_rtp_engine *current_engine = NULL;
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AST_RWLIST_WRLOCK(&engines);
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if ((current_engine = AST_RWLIST_REMOVE(&engines, engine, entry))) {
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ast_verb(2, "Unregistered RTP engine '%s'\n", engine->name);
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}
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AST_RWLIST_UNLOCK(&engines);
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return current_engine ? 0 : -1;
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}
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int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module)
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{
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struct ast_rtp_glue *current_glue = NULL;
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if (ast_strlen_zero(glue->type)) {
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return -1;
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}
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glue->mod = module;
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AST_RWLIST_WRLOCK(&glues);
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AST_RWLIST_TRAVERSE(&glues, current_glue, entry) {
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if (!strcasecmp(current_glue->type, glue->type)) {
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ast_log(LOG_WARNING, "RTP glue with the name '%s' has already been registered.\n", glue->type);
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AST_RWLIST_UNLOCK(&glues);
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return -1;
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}
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}
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AST_RWLIST_INSERT_TAIL(&glues, glue, entry);
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AST_RWLIST_UNLOCK(&glues);
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ast_verb(2, "Registered RTP glue '%s'\n", glue->type);
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return 0;
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}
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int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
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{
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struct ast_rtp_glue *current_glue = NULL;
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AST_RWLIST_WRLOCK(&glues);
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if ((current_glue = AST_RWLIST_REMOVE(&glues, glue, entry))) {
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ast_verb(2, "Unregistered RTP glue '%s'\n", glue->type);
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}
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AST_RWLIST_UNLOCK(&glues);
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return current_glue ? 0 : -1;
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}
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static void instance_destructor(void *obj)
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{
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struct ast_rtp_instance *instance = obj;
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/* Pass us off to the engine to destroy */
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if (instance->data && instance->engine->destroy(instance)) {
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ast_debug(1, "Engine '%s' failed to destroy RTP instance '%p'\n", instance->engine->name, instance);
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return;
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}
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if (instance->srtp) {
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res_srtp->destroy(instance->srtp);
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}
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ast_rtp_codecs_payloads_destroy(&instance->codecs);
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/* Drop our engine reference */
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ast_module_unref(instance->engine->mod);
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ast_debug(1, "Destroyed RTP instance '%p'\n", instance);
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}
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int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
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{
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ao2_ref(instance, -1);
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return 0;
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}
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struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
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struct ast_sched_context *sched, const struct ast_sockaddr *sa,
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void *data)
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{
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struct ast_sockaddr address = {{0,}};
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struct ast_rtp_instance *instance = NULL;
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struct ast_rtp_engine *engine = NULL;
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AST_RWLIST_RDLOCK(&engines);
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/* If an engine name was specified try to use it or otherwise use the first one registered */
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if (!ast_strlen_zero(engine_name)) {
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AST_RWLIST_TRAVERSE(&engines, engine, entry) {
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if (!strcmp(engine->name, engine_name)) {
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break;
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}
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}
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} else {
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engine = AST_RWLIST_FIRST(&engines);
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}
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/* If no engine was actually found bail out now */
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if (!engine) {
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ast_log(LOG_ERROR, "No RTP engine was found. Do you have one loaded?\n");
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AST_RWLIST_UNLOCK(&engines);
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return NULL;
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}
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/* Bump up the reference count before we return so the module can not be unloaded */
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ast_module_ref(engine->mod);
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AST_RWLIST_UNLOCK(&engines);
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/* Allocate a new RTP instance */
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if (!(instance = ao2_alloc(sizeof(*instance), instance_destructor))) {
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ast_module_unref(engine->mod);
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return NULL;
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}
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instance->engine = engine;
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ast_sockaddr_copy(&instance->local_address, sa);
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ast_sockaddr_copy(&address, sa);
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if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
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ao2_ref(instance, -1);
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return NULL;
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}
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ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
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|
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/* And pass it off to the engine to setup */
|
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if (instance->engine->new(instance, sched, &address, data)) {
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ast_debug(1, "Engine '%s' failed to setup RTP instance '%p'\n", engine->name, instance);
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ao2_ref(instance, -1);
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return NULL;
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}
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|
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ast_debug(1, "RTP instance '%p' is setup and ready to go\n", instance);
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|
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return instance;
|
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}
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|
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const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance)
|
|
{
|
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return instance->channel_uniqueid;
|
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}
|
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|
|
void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
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{
|
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ast_copy_string(instance->channel_uniqueid, uniqueid, sizeof(instance->channel_uniqueid));
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}
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|
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void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
|
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{
|
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instance->data = data;
|
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}
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|
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void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance)
|
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{
|
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return instance->data;
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}
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|
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int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
|
{
|
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return instance->engine->write(instance, frame);
|
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}
|
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|
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struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp)
|
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{
|
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return instance->engine->read(instance, rtcp);
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}
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|
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int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
|
|
const struct ast_sockaddr *address)
|
|
{
|
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ast_sockaddr_copy(&instance->local_address, address);
|
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return 0;
|
|
}
|
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|
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int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance,
|
|
const struct ast_sockaddr *address)
|
|
{
|
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ast_sockaddr_copy(&instance->remote_address, address);
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|
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/* moo */
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if (instance->engine->remote_address_set) {
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instance->engine->remote_address_set(instance, &instance->remote_address);
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}
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|
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return 0;
|
|
}
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|
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int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
|
|
struct ast_sockaddr *address)
|
|
{
|
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if (ast_sockaddr_cmp(address, &instance->local_address) != 0) {
|
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ast_sockaddr_copy(address, &instance->local_address);
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return 1;
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|
}
|
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|
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return 0;
|
|
}
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|
|
void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance,
|
|
struct ast_sockaddr *address)
|
|
{
|
|
ast_sockaddr_copy(address, &instance->local_address);
|
|
}
|
|
|
|
int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance,
|
|
struct ast_sockaddr *address)
|
|
{
|
|
if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
|
|
ast_sockaddr_copy(address, &instance->remote_address);
|
|
return 1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
|
|
struct ast_sockaddr *address)
|
|
{
|
|
ast_sockaddr_copy(address, &instance->remote_address);
|
|
}
|
|
|
|
void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
|
|
{
|
|
if (instance->engine->extended_prop_set) {
|
|
instance->engine->extended_prop_set(instance, property, value);
|
|
}
|
|
}
|
|
|
|
void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property)
|
|
{
|
|
if (instance->engine->extended_prop_get) {
|
|
return instance->engine->extended_prop_get(instance, property);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
|
|
{
|
|
instance->properties[property] = value;
|
|
|
|
if (instance->engine->prop_set) {
|
|
instance->engine->prop_set(instance, property, value);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property)
|
|
{
|
|
return instance->properties[property];
|
|
}
|
|
|
|
struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance)
|
|
{
|
|
return &instance->codecs;
|
|
}
|
|
|
|
static int rtp_payload_type_hash(const void *obj, const int flags)
|
|
{
|
|
const struct ast_rtp_payload_type *type = obj;
|
|
const int *payload = obj;
|
|
|
|
return (flags & OBJ_KEY) ? *payload : type->payload;
|
|
}
|
|
|
|
static int rtp_payload_type_cmp(void *obj, void *arg, int flags)
|
|
{
|
|
struct ast_rtp_payload_type *type1 = obj, *type2 = arg;
|
|
const int *payload = arg;
|
|
|
|
return (type1->payload == (OBJ_KEY ? *payload : type2->payload)) ? CMP_MATCH | CMP_STOP : 0;
|
|
}
|
|
|
|
int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
|
|
{
|
|
if (!(codecs->payloads = ao2_container_alloc(AST_RTP_MAX_PT, rtp_payload_type_hash, rtp_payload_type_cmp))) {
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
|
|
{
|
|
ao2_cleanup(codecs->payloads);
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
|
|
{
|
|
ast_rtp_codecs_payloads_destroy(codecs);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
int i;
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
instance->engine->payload_set(instance, i, 0, NULL, 0);
|
|
}
|
|
}
|
|
|
|
ast_rtp_codecs_payloads_initialize(codecs);
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
|
|
{
|
|
int i;
|
|
|
|
ast_rwlock_rdlock(&static_RTP_PT_lock);
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
|
|
struct ast_rtp_payload_type *type;
|
|
|
|
if (!(type = ao2_alloc(sizeof(*type), NULL))) {
|
|
/* Unfortunately if this occurs the payloads container will not contain all possible default payloads
|
|
* but we err on the side of doing what we can in the hopes that the extreme memory conditions which
|
|
* caused this to occur will go away.
|
|
*/
|
|
continue;
|
|
}
|
|
|
|
type->payload = i;
|
|
type->asterisk_format = static_RTP_PT[i].asterisk_format;
|
|
type->rtp_code = static_RTP_PT[i].rtp_code;
|
|
ast_format_copy(&type->format, &static_RTP_PT[i].format);
|
|
|
|
ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
|
|
}
|
|
|
|
ao2_ref(type, -1);
|
|
}
|
|
}
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
|
|
{
|
|
int i;
|
|
struct ast_rtp_payload_type *type;
|
|
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
struct ast_rtp_payload_type *new_type;
|
|
|
|
if (!(type = ao2_find(src->payloads, &i, OBJ_KEY | OBJ_NOLOCK))) {
|
|
continue;
|
|
}
|
|
|
|
if (!(new_type = ao2_alloc(sizeof(*new_type), NULL))) {
|
|
continue;
|
|
}
|
|
|
|
ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
|
|
|
|
new_type->payload = i;
|
|
*new_type = *type;
|
|
|
|
ao2_link_flags(dest->payloads, new_type, OBJ_NOLOCK);
|
|
|
|
ao2_ref(new_type, -1);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, i, type->asterisk_format, &type->format, type->rtp_code);
|
|
}
|
|
|
|
ao2_ref(type, -1);
|
|
}
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
|
|
{
|
|
struct ast_rtp_payload_type *type;
|
|
|
|
ast_rwlock_rdlock(&static_RTP_PT_lock);
|
|
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
return;
|
|
}
|
|
|
|
if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
|
|
if (!(type = ao2_alloc(sizeof(*type), NULL))) {
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
return;
|
|
}
|
|
type->payload = payload;
|
|
ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
|
|
}
|
|
|
|
type->asterisk_format = static_RTP_PT[payload].asterisk_format;
|
|
type->rtp_code = static_RTP_PT[payload].rtp_code;
|
|
type->payload = payload;
|
|
ast_format_copy(&type->format, &static_RTP_PT[payload].format);
|
|
|
|
ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, payload, type->asterisk_format, &type->format, type->rtp_code);
|
|
}
|
|
|
|
ao2_ref(type, -1);
|
|
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
}
|
|
|
|
int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
|
|
char *mimetype, char *mimesubtype,
|
|
enum ast_rtp_options options,
|
|
unsigned int sample_rate)
|
|
{
|
|
unsigned int i;
|
|
int found = 0;
|
|
|
|
if (pt < 0 || pt >= AST_RTP_MAX_PT)
|
|
return -1; /* bogus payload type */
|
|
|
|
ast_rwlock_rdlock(&mime_types_lock);
|
|
for (i = 0; i < mime_types_len; ++i) {
|
|
const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
|
|
struct ast_rtp_payload_type *type;
|
|
|
|
if (strcasecmp(mimesubtype, t->subtype)) {
|
|
continue;
|
|
}
|
|
|
|
if (strcasecmp(mimetype, t->type)) {
|
|
continue;
|
|
}
|
|
|
|
/* if both sample rates have been supplied, and they don't match,
|
|
* then this not a match; if one has not been supplied, then the
|
|
* rates are not compared */
|
|
if (sample_rate && t->sample_rate &&
|
|
(sample_rate != t->sample_rate)) {
|
|
continue;
|
|
}
|
|
|
|
found = 1;
|
|
|
|
if (!(type = ao2_find(codecs->payloads, &pt, OBJ_KEY | OBJ_NOLOCK))) {
|
|
if (!(type = ao2_alloc(sizeof(*type), NULL))) {
|
|
continue;
|
|
}
|
|
type->payload = pt;
|
|
ao2_link_flags(codecs->payloads, type, OBJ_NOLOCK);
|
|
}
|
|
|
|
*type = t->payload_type;
|
|
type->payload = pt;
|
|
|
|
if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
|
|
ast_format_set(&type->format, AST_FORMAT_G726_AAL2, 0);
|
|
}
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, pt, type->asterisk_format, &type->format, type->rtp_code);
|
|
}
|
|
|
|
ao2_ref(type, -1);
|
|
|
|
break;
|
|
}
|
|
ast_rwlock_unlock(&mime_types_lock);
|
|
|
|
return (found ? 0 : -2);
|
|
}
|
|
|
|
int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options)
|
|
{
|
|
return ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, instance, payload, mimetype, mimesubtype, options, 0);
|
|
}
|
|
|
|
void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
|
|
{
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
return;
|
|
}
|
|
|
|
ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
|
|
|
|
ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK | OBJ_NODATA | OBJ_UNLINK);
|
|
|
|
if (instance && instance->engine && instance->engine->payload_set) {
|
|
instance->engine->payload_set(instance, payload, 0, NULL, 0);
|
|
}
|
|
}
|
|
|
|
struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
|
|
{
|
|
struct ast_rtp_payload_type result = { .asterisk_format = 0, }, *type;
|
|
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
return result;
|
|
}
|
|
|
|
if ((type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
|
|
result = *type;
|
|
ao2_ref(type, -1);
|
|
}
|
|
|
|
if (!result.rtp_code && !result.asterisk_format) {
|
|
ast_rwlock_rdlock(&static_RTP_PT_lock);
|
|
result = static_RTP_PT[payload];
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
|
|
struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
|
|
{
|
|
struct ast_rtp_payload_type *type;
|
|
struct ast_format *format;
|
|
|
|
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
|
|
return NULL;
|
|
}
|
|
|
|
if (!(type = ao2_find(codecs->payloads, &payload, OBJ_KEY | OBJ_NOLOCK))) {
|
|
return NULL;
|
|
}
|
|
|
|
format = type->asterisk_format ? &type->format : NULL;
|
|
|
|
ao2_ref(type, -1);
|
|
|
|
return format;
|
|
}
|
|
|
|
static int rtp_payload_type_add_ast(void *obj, void *arg, int flags)
|
|
{
|
|
struct ast_rtp_payload_type *type = obj;
|
|
struct ast_format_cap *astformats = arg;
|
|
|
|
if (type->asterisk_format) {
|
|
ast_format_cap_add(astformats, &type->format);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_payload_type_add_nonast(void *obj, void *arg, int flags)
|
|
{
|
|
struct ast_rtp_payload_type *type = obj;
|
|
int *nonastformats = arg;
|
|
|
|
if (!type->asterisk_format) {
|
|
*nonastformats |= type->rtp_code;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
|
|
{
|
|
ast_format_cap_remove_all(astformats);
|
|
*nonastformats = 0;
|
|
|
|
ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_ast, astformats);
|
|
ao2_callback(codecs->payloads, OBJ_NODATA | OBJ_MULTIPLE | OBJ_NOLOCK, rtp_payload_type_add_nonast, nonastformats);
|
|
}
|
|
|
|
static int rtp_payload_type_find_format(void *obj, void *arg, int flags)
|
|
{
|
|
struct ast_rtp_payload_type *type = obj;
|
|
struct ast_format *format = arg;
|
|
|
|
return (type->asterisk_format && (ast_format_cmp(&type->format, format) != AST_FORMAT_CMP_NOT_EQUAL)) ? CMP_MATCH | CMP_STOP : 0;
|
|
}
|
|
|
|
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
|
|
{
|
|
struct ast_rtp_payload_type *type;
|
|
int i, res = -1;
|
|
|
|
if (asterisk_format && format && (type = ao2_callback(codecs->payloads, OBJ_NOLOCK, rtp_payload_type_find_format, (void*)format))) {
|
|
res = type->payload;
|
|
ao2_ref(type, -1);
|
|
return res;
|
|
} else if (!asterisk_format && (type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY))) {
|
|
res = type->payload;
|
|
ao2_ref(type, -1);
|
|
return res;
|
|
}
|
|
|
|
ast_rwlock_rdlock(&static_RTP_PT_lock);
|
|
for (i = 0; i < AST_RTP_MAX_PT; i++) {
|
|
if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
|
|
(ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
|
|
res = i;
|
|
break;
|
|
} else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
|
|
(static_RTP_PT[i].rtp_code == code)) {
|
|
res = i;
|
|
break;
|
|
}
|
|
}
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
|
|
return res;
|
|
}
|
|
int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int code)
|
|
{
|
|
struct ast_rtp_payload_type *type;
|
|
int res = -1;
|
|
|
|
/* Search the payload type in the codecs passed */
|
|
if ((type = ao2_find(codecs->payloads, &code, OBJ_NOLOCK | OBJ_KEY)))
|
|
{
|
|
res = type->payload;
|
|
ao2_ref(type, -1);
|
|
return res;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
|
|
{
|
|
int i;
|
|
const char *res = "";
|
|
|
|
ast_rwlock_rdlock(&mime_types_lock);
|
|
for (i = 0; i < mime_types_len; i++) {
|
|
if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
|
|
(ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
|
|
if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
|
|
res = "G726-32";
|
|
break;
|
|
} else {
|
|
res = ast_rtp_mime_types[i].subtype;
|
|
break;
|
|
}
|
|
} else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
|
|
ast_rtp_mime_types[i].payload_type.rtp_code == code) {
|
|
|
|
res = ast_rtp_mime_types[i].subtype;
|
|
break;
|
|
}
|
|
}
|
|
ast_rwlock_unlock(&mime_types_lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
|
|
{
|
|
unsigned int i;
|
|
unsigned int res = 0;
|
|
|
|
ast_rwlock_rdlock(&mime_types_lock);
|
|
for (i = 0; i < mime_types_len; ++i) {
|
|
if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
|
|
(ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
|
|
res = ast_rtp_mime_types[i].sample_rate;
|
|
break;
|
|
} else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
|
|
ast_rtp_mime_types[i].payload_type.rtp_code == code) {
|
|
res = ast_rtp_mime_types[i].sample_rate;
|
|
break;
|
|
}
|
|
}
|
|
ast_rwlock_unlock(&mime_types_lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, struct ast_format_cap *ast_format_capability, int rtp_capability, const int asterisk_format, enum ast_rtp_options options)
|
|
{
|
|
int found = 0;
|
|
const char *name;
|
|
if (!buf) {
|
|
return NULL;
|
|
}
|
|
|
|
|
|
if (asterisk_format) {
|
|
struct ast_format tmp_fmt;
|
|
ast_format_cap_iter_start(ast_format_capability);
|
|
while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
|
|
name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
|
|
ast_str_append(&buf, 0, "%s|", name);
|
|
found = 1;
|
|
}
|
|
ast_format_cap_iter_end(ast_format_capability);
|
|
|
|
} else {
|
|
int x;
|
|
ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
|
|
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
|
|
if (rtp_capability & x) {
|
|
name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
|
|
ast_str_append(&buf, 0, "%s|", name);
|
|
found = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
ast_str_append(&buf, 0, "%s", found ? ")" : "nothing)");
|
|
|
|
return ast_str_buffer(buf);
|
|
}
|
|
|
|
void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
|
|
{
|
|
codecs->pref = *prefs;
|
|
|
|
if (instance && instance->engine->packetization_set) {
|
|
instance->engine->packetization_set(instance, &instance->codecs.pref);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
return instance->engine->dtmf_end ? instance->engine->dtmf_end(instance, digit) : -1;
|
|
}
|
|
int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
|
|
{
|
|
return instance->engine->dtmf_end_with_duration ? instance->engine->dtmf_end_with_duration(instance, digit, duration) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
|
|
{
|
|
return (!instance->engine->dtmf_mode_set || instance->engine->dtmf_mode_set(instance, dtmf_mode)) ? -1 : 0;
|
|
}
|
|
|
|
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine->dtmf_mode_get ? instance->engine->dtmf_mode_get(instance) : 0;
|
|
}
|
|
|
|
void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
|
|
{
|
|
if (instance->engine->update_source) {
|
|
instance->engine->update_source(instance);
|
|
}
|
|
}
|
|
|
|
void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
|
|
{
|
|
if (instance->engine->change_source) {
|
|
instance->engine->change_source(instance);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
|
|
{
|
|
return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
|
|
}
|
|
|
|
void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
|
|
{
|
|
if (instance->engine->stop) {
|
|
instance->engine->stop(instance);
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
return instance->engine->fd ? instance->engine->fd(instance, rtcp) : -1;
|
|
}
|
|
|
|
struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type)
|
|
{
|
|
struct ast_rtp_glue *glue = NULL;
|
|
|
|
AST_RWLIST_RDLOCK(&glues);
|
|
|
|
AST_RWLIST_TRAVERSE(&glues, glue, entry) {
|
|
if (!strcasecmp(glue->type, type)) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
AST_RWLIST_UNLOCK(&glues);
|
|
|
|
return glue;
|
|
}
|
|
|
|
/*!
|
|
* \brief Conditionally unref an rtp instance
|
|
*/
|
|
static void unref_instance_cond(struct ast_rtp_instance **instance)
|
|
{
|
|
if (*instance) {
|
|
ao2_ref(*instance, -1);
|
|
*instance = NULL;
|
|
}
|
|
}
|
|
|
|
struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->bridged;
|
|
}
|
|
|
|
void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
|
|
{
|
|
instance->bridged = bridged;
|
|
}
|
|
|
|
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
|
|
{
|
|
struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
|
|
*vinstance0 = NULL, *vinstance1 = NULL,
|
|
*tinstance0 = NULL, *tinstance1 = NULL;
|
|
struct ast_rtp_glue *glue0, *glue1;
|
|
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
|
|
struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
|
|
|
|
/* Lock both channels so we can look for the glue that binds them together */
|
|
ast_channel_lock_both(c0, c1);
|
|
|
|
if (!cap1 || !cap0) {
|
|
goto done;
|
|
}
|
|
|
|
/* Grab glue that binds each channel to something using the RTP engine */
|
|
if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
|
|
ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
|
|
goto done;
|
|
}
|
|
|
|
audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
|
|
video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
|
|
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
|
|
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) {
|
|
glue0->get_codec(c0, cap0);
|
|
}
|
|
if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec) {
|
|
glue1->get_codec(c1, cap1);
|
|
}
|
|
|
|
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
|
|
if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
|
|
goto done;
|
|
}
|
|
|
|
/* Make sure we have matching codecs */
|
|
if (!ast_format_cap_has_joint(cap0, cap1)) {
|
|
goto done;
|
|
}
|
|
|
|
ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
|
|
|
|
if (vinstance0 && vinstance1) {
|
|
ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
|
|
}
|
|
if (tinstance0 && tinstance1) {
|
|
ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
|
|
}
|
|
|
|
if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
|
|
ast_channel_name(c0), ast_channel_name(c1));
|
|
} else {
|
|
ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
|
|
ast_channel_name(c0), ast_channel_name(c1));
|
|
}
|
|
|
|
done:
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
ast_format_cap_destroy(cap0);
|
|
ast_format_cap_destroy(cap1);
|
|
|
|
unref_instance_cond(&instance0);
|
|
unref_instance_cond(&instance1);
|
|
unref_instance_cond(&vinstance0);
|
|
unref_instance_cond(&vinstance1);
|
|
unref_instance_cond(&tinstance0);
|
|
unref_instance_cond(&tinstance1);
|
|
}
|
|
|
|
int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
|
|
{
|
|
struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
|
|
*vinstance0 = NULL, *vinstance1 = NULL,
|
|
*tinstance0 = NULL, *tinstance1 = NULL;
|
|
struct ast_rtp_glue *glue0, *glue1;
|
|
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
|
|
struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
|
|
|
|
/* If there is no second channel just immediately bail out, we are of no use in that scenario */
|
|
if (!c1 || !cap1 || !cap0) {
|
|
ast_format_cap_destroy(cap0);
|
|
ast_format_cap_destroy(cap1);
|
|
return -1;
|
|
}
|
|
|
|
/* Lock both channels so we can look for the glue that binds them together */
|
|
ast_channel_lock_both(c0, c1);
|
|
|
|
/* Grab glue that binds each channel to something using the RTP engine */
|
|
if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
|
|
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
|
|
goto done;
|
|
}
|
|
|
|
audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
|
|
video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
|
|
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
|
|
|
|
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
|
|
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
|
|
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
|
|
}
|
|
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) {
|
|
glue0->get_codec(c0, cap0);
|
|
}
|
|
if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec) {
|
|
glue1->get_codec(c1, cap1);
|
|
}
|
|
|
|
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
|
|
if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
|
|
goto done;
|
|
}
|
|
|
|
/* Make sure we have matching codecs */
|
|
if (!ast_format_cap_has_joint(cap0, cap1)) {
|
|
goto done;
|
|
}
|
|
|
|
/* Bridge media early */
|
|
if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
|
|
}
|
|
|
|
done:
|
|
ast_channel_unlock(c0);
|
|
ast_channel_unlock(c1);
|
|
|
|
ast_format_cap_destroy(cap0);
|
|
ast_format_cap_destroy(cap1);
|
|
|
|
unref_instance_cond(&instance0);
|
|
unref_instance_cond(&instance1);
|
|
unref_instance_cond(&vinstance0);
|
|
unref_instance_cond(&vinstance1);
|
|
unref_instance_cond(&tinstance0);
|
|
unref_instance_cond(&tinstance1);
|
|
|
|
ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
|
|
{
|
|
return instance->engine->red_init ? instance->engine->red_init(instance, buffer_time, payloads, generations) : -1;
|
|
}
|
|
|
|
int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
|
{
|
|
return instance->engine->red_buffer ? instance->engine->red_buffer(instance, frame) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
|
|
{
|
|
return instance->engine->get_stat ? instance->engine->get_stat(instance, stats, stat) : -1;
|
|
}
|
|
|
|
char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size)
|
|
{
|
|
struct ast_rtp_instance_stats stats = { 0, };
|
|
enum ast_rtp_instance_stat stat;
|
|
|
|
/* Determine what statistics we will need to retrieve based on field passed in */
|
|
if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
|
|
stat = AST_RTP_INSTANCE_STAT_ALL;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_JITTER;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_LOSS;
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
|
|
stat = AST_RTP_INSTANCE_STAT_COMBINED_RTT;
|
|
} else {
|
|
return NULL;
|
|
}
|
|
|
|
/* Attempt to actually retrieve the statistics we need to generate the quality string */
|
|
if (ast_rtp_instance_get_stats(instance, &stats, stat)) {
|
|
return NULL;
|
|
}
|
|
|
|
/* Now actually fill the buffer with the good information */
|
|
if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
|
|
snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
|
|
stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
|
|
snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
|
|
stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS) {
|
|
snprintf(buf, size, "minrxlost=%f;maxrxlost=%f;avgrxlost=%f;stdevrxlost=%f;reported_minlost=%f;reported_maxlost=%f;reported_avglost=%f;reported_stdevlost=%f;",
|
|
stats.local_minrxploss, stats.local_maxrxploss, stats.local_normdevrxploss, sqrt(stats.local_stdevrxploss), stats.remote_minrxploss, stats.remote_maxrxploss, stats.remote_normdevrxploss, sqrt(stats.remote_stdevrxploss));
|
|
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT) {
|
|
snprintf(buf, size, "minrtt=%f;maxrtt=%f;avgrtt=%f;stdevrtt=%f;", stats.minrtt, stats.maxrtt, stats.normdevrtt, stats.stdevrtt);
|
|
}
|
|
|
|
return buf;
|
|
}
|
|
|
|
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
|
|
{
|
|
char quality_buf[AST_MAX_USER_FIELD], *quality;
|
|
RAII_VAR(struct ast_channel *, bridge, ast_channel_bridge_peer(chan), ast_channel_cleanup);
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
|
|
}
|
|
}
|
|
|
|
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
|
|
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
|
|
if (bridge) {
|
|
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
|
|
}
|
|
}
|
|
}
|
|
|
|
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
|
|
{
|
|
return instance->engine->set_read_format ? instance->engine->set_read_format(instance, format) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format)
|
|
{
|
|
return instance->engine->set_write_format ? instance->engine->set_write_format(instance, format) : -1;
|
|
}
|
|
|
|
int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer)
|
|
{
|
|
struct ast_rtp_glue *glue;
|
|
struct ast_rtp_instance *peer_instance = NULL;
|
|
int res = -1;
|
|
|
|
if (!instance->engine->make_compatible) {
|
|
return -1;
|
|
}
|
|
|
|
ast_channel_lock(peer);
|
|
|
|
if (!(glue = ast_rtp_instance_get_glue(ast_channel_tech(peer)->type))) {
|
|
ast_channel_unlock(peer);
|
|
return -1;
|
|
}
|
|
|
|
glue->get_rtp_info(peer, &peer_instance);
|
|
|
|
if (!peer_instance || peer_instance->engine != instance->engine) {
|
|
ast_channel_unlock(peer);
|
|
ao2_ref(peer_instance, -1);
|
|
peer_instance = NULL;
|
|
return -1;
|
|
}
|
|
|
|
res = instance->engine->make_compatible(chan, instance, peer, peer_instance);
|
|
|
|
ast_channel_unlock(peer);
|
|
|
|
ao2_ref(peer_instance, -1);
|
|
peer_instance = NULL;
|
|
|
|
return res;
|
|
}
|
|
|
|
void ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result)
|
|
{
|
|
if (instance->engine->available_formats) {
|
|
instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
|
|
if (!ast_format_cap_is_empty(result)) {
|
|
return;
|
|
}
|
|
}
|
|
|
|
ast_translate_available_formats(to_endpoint, to_asterisk, result);
|
|
}
|
|
|
|
int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine->activate ? instance->engine->activate(instance) : 0;
|
|
}
|
|
|
|
void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance,
|
|
struct ast_sockaddr *suggestion,
|
|
const char *username)
|
|
{
|
|
if (instance->engine->stun_request) {
|
|
instance->engine->stun_request(instance, suggestion, username);
|
|
}
|
|
}
|
|
|
|
void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout)
|
|
{
|
|
instance->timeout = timeout;
|
|
}
|
|
|
|
void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout)
|
|
{
|
|
instance->holdtimeout = timeout;
|
|
}
|
|
|
|
void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
|
|
{
|
|
instance->keepalive = interval;
|
|
}
|
|
|
|
int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->timeout;
|
|
}
|
|
|
|
int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->holdtimeout;
|
|
}
|
|
|
|
int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->keepalive;
|
|
}
|
|
|
|
struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine;
|
|
}
|
|
|
|
struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->glue;
|
|
}
|
|
|
|
int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
|
|
{
|
|
if (res_srtp || res_srtp_policy) {
|
|
return -1;
|
|
}
|
|
if (!srtp_res || !policy_res) {
|
|
return -1;
|
|
}
|
|
|
|
res_srtp = srtp_res;
|
|
res_srtp_policy = policy_res;
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_rtp_engine_unregister_srtp(void)
|
|
{
|
|
res_srtp = NULL;
|
|
res_srtp_policy = NULL;
|
|
}
|
|
|
|
int ast_rtp_engine_srtp_is_registered(void)
|
|
{
|
|
return res_srtp && res_srtp_policy;
|
|
}
|
|
|
|
int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
|
|
{
|
|
int res = 0;
|
|
|
|
if (!res_srtp) {
|
|
return -1;
|
|
}
|
|
|
|
if (!instance->srtp) {
|
|
res = res_srtp->create(&instance->srtp, instance, remote_policy);
|
|
} else {
|
|
res = res_srtp->replace(&instance->srtp, instance, remote_policy);
|
|
}
|
|
if (!res) {
|
|
res = res_srtp->add_stream(instance->srtp, local_policy);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->srtp;
|
|
}
|
|
|
|
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
|
|
{
|
|
if (instance->engine->sendcng) {
|
|
return instance->engine->sendcng(instance, level);
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine->ice;
|
|
}
|
|
|
|
struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance)
|
|
{
|
|
return instance->engine->dtls;
|
|
}
|
|
|
|
int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value)
|
|
{
|
|
if (!strcasecmp(name, "dtlsenable")) {
|
|
dtls_cfg->enabled = ast_true(value) ? 1 : 0;
|
|
} else if (!strcasecmp(name, "dtlsverify")) {
|
|
dtls_cfg->verify = ast_true(value) ? 1 : 0;
|
|
} else if (!strcasecmp(name, "dtlsrekey")) {
|
|
if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
|
|
return -1;
|
|
}
|
|
} else if (!strcasecmp(name, "dtlscertfile")) {
|
|
ast_free(dtls_cfg->certfile);
|
|
dtls_cfg->certfile = ast_strdup(value);
|
|
} else if (!strcasecmp(name, "dtlsprivatekey")) {
|
|
ast_free(dtls_cfg->pvtfile);
|
|
dtls_cfg->pvtfile = ast_strdup(value);
|
|
} else if (!strcasecmp(name, "dtlscipher")) {
|
|
ast_free(dtls_cfg->cipher);
|
|
dtls_cfg->cipher = ast_strdup(value);
|
|
} else if (!strcasecmp(name, "dtlscafile")) {
|
|
ast_free(dtls_cfg->cafile);
|
|
dtls_cfg->cafile = ast_strdup(value);
|
|
} else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
|
|
ast_free(dtls_cfg->capath);
|
|
dtls_cfg->capath = ast_strdup(value);
|
|
} else if (!strcasecmp(name, "dtlssetup")) {
|
|
if (!strcasecmp(value, "active")) {
|
|
dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
|
|
} else if (!strcasecmp(value, "passive")) {
|
|
dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
|
|
} else if (!strcasecmp(value, "actpass")) {
|
|
dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
|
|
}
|
|
} else {
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg)
|
|
{
|
|
dst_cfg->enabled = src_cfg->enabled;
|
|
dst_cfg->verify = src_cfg->verify;
|
|
dst_cfg->rekey = src_cfg->rekey;
|
|
dst_cfg->suite = src_cfg->suite;
|
|
dst_cfg->certfile = ast_strdup(src_cfg->certfile);
|
|
dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
|
|
dst_cfg->cipher = ast_strdup(src_cfg->cipher);
|
|
dst_cfg->cafile = ast_strdup(src_cfg->cafile);
|
|
dst_cfg->capath = ast_strdup(src_cfg->capath);
|
|
dst_cfg->default_setup = src_cfg->default_setup;
|
|
}
|
|
|
|
void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg)
|
|
{
|
|
ast_free(dtls_cfg->certfile);
|
|
ast_free(dtls_cfg->pvtfile);
|
|
ast_free(dtls_cfg->cipher);
|
|
ast_free(dtls_cfg->cafile);
|
|
ast_free(dtls_cfg->capath);
|
|
}
|
|
|
|
static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
|
|
{
|
|
int x = mime_types_len;
|
|
if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
|
|
return;
|
|
}
|
|
|
|
ast_rwlock_wrlock(&mime_types_lock);
|
|
if (format) {
|
|
ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
|
|
ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
|
|
} else {
|
|
ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
|
|
}
|
|
ast_rtp_mime_types[x].type = type;
|
|
ast_rtp_mime_types[x].subtype = subtype;
|
|
ast_rtp_mime_types[x].sample_rate = sample_rate;
|
|
mime_types_len++;
|
|
ast_rwlock_unlock(&mime_types_lock);
|
|
}
|
|
|
|
static void add_static_payload(int map, const struct ast_format *format, int rtp_code)
|
|
{
|
|
int x;
|
|
ast_rwlock_wrlock(&static_RTP_PT_lock);
|
|
if (map < 0) {
|
|
/* find next available dynamic payload slot */
|
|
for (x = 96; x < 127; x++) {
|
|
if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
|
|
map = x;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (map < 0) {
|
|
ast_log(LOG_WARNING, "No Dynamic RTP mapping avaliable for format %s\n" ,ast_getformatname(format));
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
return;
|
|
}
|
|
|
|
if (format) {
|
|
static_RTP_PT[map].asterisk_format = 1;
|
|
ast_format_copy(&static_RTP_PT[map].format, format);
|
|
} else {
|
|
static_RTP_PT[map].rtp_code = rtp_code;
|
|
}
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
}
|
|
|
|
int ast_rtp_engine_load_format(const struct ast_format *format)
|
|
{
|
|
switch (format->id) {
|
|
case AST_FORMAT_SILK:
|
|
set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
|
|
add_static_payload(-1, format, 0);
|
|
break;
|
|
case AST_FORMAT_CELT:
|
|
set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
|
|
add_static_payload(-1, format, 0);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_rtp_engine_unload_format(const struct ast_format *format)
|
|
{
|
|
int x;
|
|
int y = 0;
|
|
|
|
ast_rwlock_wrlock(&static_RTP_PT_lock);
|
|
/* remove everything pertaining to this format id from the lists */
|
|
for (x = 0; x < AST_RTP_MAX_PT; x++) {
|
|
if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
|
|
memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
|
|
}
|
|
}
|
|
ast_rwlock_unlock(&static_RTP_PT_lock);
|
|
|
|
|
|
ast_rwlock_wrlock(&mime_types_lock);
|
|
/* rebuild the list skipping the items matching this id */
|
|
for (x = 0; x < mime_types_len; x++) {
|
|
if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
|
|
continue;
|
|
}
|
|
ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
|
|
y++;
|
|
}
|
|
mime_types_len = y;
|
|
ast_rwlock_unlock(&mime_types_lock);
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief \ref stasis message payload for RTCP messages
|
|
*/
|
|
struct rtcp_message_payload {
|
|
struct ast_channel_snapshot *snapshot; /*< The channel snapshot, if available */
|
|
struct ast_rtp_rtcp_report *report; /*< The RTCP report */
|
|
struct ast_json *blob; /*< Extra JSON data to publish */
|
|
};
|
|
|
|
static void rtcp_message_payload_dtor(void *obj)
|
|
{
|
|
struct rtcp_message_payload *payload = obj;
|
|
|
|
ao2_cleanup(payload->report);
|
|
ao2_cleanup(payload->snapshot);
|
|
ast_json_unref(payload->blob);
|
|
}
|
|
|
|
static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *msg)
|
|
{
|
|
struct rtcp_message_payload *payload = stasis_message_data(msg);
|
|
RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
|
|
RAII_VAR(struct ast_str *, packet_string, ast_str_create(512), ast_free);
|
|
unsigned int ssrc = payload->report->ssrc;
|
|
unsigned int type = payload->report->type;
|
|
unsigned int report_count = payload->report->reception_report_count;
|
|
int i;
|
|
|
|
if (!packet_string) {
|
|
return NULL;
|
|
}
|
|
|
|
if (payload->snapshot) {
|
|
channel_string = ast_manager_build_channel_state_string(payload->snapshot);
|
|
if (!channel_string) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
if (payload->blob) {
|
|
/* Optional data */
|
|
struct ast_json *to = ast_json_object_get(payload->blob, "to");
|
|
struct ast_json *from = ast_json_object_get(payload->blob, "from");
|
|
struct ast_json *rtt = ast_json_object_get(payload->blob, "rtt");
|
|
if (to) {
|
|
ast_str_append(&packet_string, 0, "To: %s\r\n", ast_json_string_get(to));
|
|
}
|
|
if (from) {
|
|
ast_str_append(&packet_string, 0, "From: %s\r\n", ast_json_string_get(from));
|
|
}
|
|
if (rtt) {
|
|
ast_str_append(&packet_string, 0, "RTT: %4.4f\r\n", ast_json_real_get(rtt));
|
|
}
|
|
}
|
|
|
|
ast_str_append(&packet_string, 0, "SSRC: 0x%.8x\r\n", ssrc);
|
|
ast_str_append(&packet_string, 0, "PT: %u(%s)\r\n", type, type== AST_RTP_RTCP_SR ? "SR" : "RR");
|
|
ast_str_append(&packet_string, 0, "ReportCount: %u\r\n", report_count);
|
|
if (type == AST_RTP_RTCP_SR) {
|
|
ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
|
|
(unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
|
|
(unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec * 4096);
|
|
ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
|
|
payload->report->sender_information.rtp_timestamp);
|
|
ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",
|
|
payload->report->sender_information.packet_count);
|
|
ast_str_append(&packet_string, 0, "SentOctets: %u\r\n",
|
|
payload->report->sender_information.octet_count);
|
|
}
|
|
|
|
for (i = 0; i < report_count; i++) {
|
|
RAII_VAR(struct ast_str *, report_string, NULL, ast_free);
|
|
|
|
if (!payload->report->report_block[i]) {
|
|
break;
|
|
}
|
|
|
|
report_string = ast_str_create(256);
|
|
if (!report_string) {
|
|
return NULL;
|
|
}
|
|
|
|
ast_str_append(&report_string, 0, "Report%dSourceSSRC: 0x%.8x\r\n",
|
|
i, payload->report->report_block[i]->source_ssrc);
|
|
ast_str_append(&report_string, 0, "Report%dFractionLost: %u\r\n",
|
|
i, payload->report->report_block[i]->lost_count.fraction);
|
|
ast_str_append(&report_string, 0, "Report%dCumulativeLost: %u\r\n",
|
|
i, payload->report->report_block[i]->lost_count.packets);
|
|
ast_str_append(&report_string, 0, "Report%dHighestSequence: %u\r\n",
|
|
i, payload->report->report_block[i]->highest_seq_no & 0xffff);
|
|
ast_str_append(&report_string, 0, "Report%dSequenceNumberCycles: %u\r\n",
|
|
i, payload->report->report_block[i]->highest_seq_no >> 16);
|
|
ast_str_append(&report_string, 0, "Report%dIAJitter: %u\r\n",
|
|
i, payload->report->report_block[i]->ia_jitter);
|
|
ast_str_append(&report_string, 0, "Report%dLSR: %u\r\n",
|
|
i, payload->report->report_block[i]->lsr);
|
|
ast_str_append(&report_string, 0, "Report%dDLSR: %4.4f\r\n",
|
|
i, ((double)payload->report->report_block[i]->dlsr) / 65536);
|
|
ast_str_append(&packet_string, 0, "%s", ast_str_buffer(report_string));
|
|
}
|
|
|
|
return ast_manager_event_blob_create(EVENT_FLAG_REPORTING,
|
|
stasis_message_type(msg) == ast_rtp_rtcp_received_type() ? "RTCPReceived" : "RTCPSent",
|
|
"%s%s",
|
|
AS_OR(channel_string, ""),
|
|
ast_str_buffer(packet_string));
|
|
}
|
|
|
|
static struct ast_json *rtcp_report_to_json(struct stasis_message *msg)
|
|
{
|
|
struct rtcp_message_payload *payload = stasis_message_data(msg);
|
|
RAII_VAR(struct ast_json *, json_rtcp_report, NULL, ast_json_unref);
|
|
RAII_VAR(struct ast_json *, json_rtcp_report_blocks, NULL, ast_json_unref);
|
|
RAII_VAR(struct ast_json *, json_rtcp_sender_info, NULL, ast_json_unref);
|
|
struct ast_json * json_payload;
|
|
int i;
|
|
|
|
json_rtcp_report_blocks = ast_json_array_create();
|
|
if (!json_rtcp_report_blocks) {
|
|
return NULL;
|
|
}
|
|
|
|
for (i = 0; i < payload->report->reception_report_count; i++) {
|
|
struct ast_json *json_report_block;
|
|
json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: i, s: i}",
|
|
"source_ssrc", payload->report->report_block[i]->source_ssrc,
|
|
"fraction_lost", payload->report->report_block[i]->lost_count.fraction,
|
|
"packets_lost", payload->report->report_block[i]->lost_count.packets,
|
|
"highest_seq_no", payload->report->report_block[i]->highest_seq_no,
|
|
"ia_jitter", payload->report->report_block[i]->ia_jitter,
|
|
"lsr", payload->report->report_block[i]->lsr,
|
|
"dlsr", payload->report->report_block[i]->dlsr);
|
|
if (!json_report_block) {
|
|
return NULL;
|
|
}
|
|
|
|
if (ast_json_array_append(json_rtcp_report_blocks, json_report_block)) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
if (payload->report->type == AST_RTP_RTCP_SR) {
|
|
json_rtcp_sender_info = ast_json_pack("{s: i, s: i, s: i, s: i, s: i}",
|
|
"ntp_timestamp_sec", payload->report->sender_information.ntp_timestamp.tv_sec,
|
|
"ntp_timestamp_usec", payload->report->sender_information.ntp_timestamp.tv_usec,
|
|
"rtp_timestamp", payload->report->sender_information.rtp_timestamp,
|
|
"packets", payload->report->sender_information.packet_count,
|
|
"octets", payload->report->sender_information.octet_count);
|
|
if (!json_rtcp_sender_info) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
json_rtcp_report = ast_json_pack("{s: i, s: i, s: i, s: O, s: O}",
|
|
"ssrc", payload->report->ssrc,
|
|
"type", payload->report->type,
|
|
"report_count", payload->report->reception_report_count,
|
|
"sender_information", json_rtcp_sender_info ? json_rtcp_sender_info : ast_json_null(),
|
|
"report_blocks", json_rtcp_report_blocks);
|
|
if (!json_rtcp_report) {
|
|
return NULL;
|
|
}
|
|
|
|
json_payload = ast_json_pack("{s: O, s: O, s: O}",
|
|
"channel", payload->snapshot ? ast_channel_snapshot_to_json(payload->snapshot) : ast_json_null(),
|
|
"rtcp_report", json_rtcp_report,
|
|
"blob", payload->blob);
|
|
return json_payload;
|
|
}
|
|
|
|
static void rtp_rtcp_report_dtor(void *obj)
|
|
{
|
|
int i;
|
|
struct ast_rtp_rtcp_report *rtcp_report = obj;
|
|
|
|
for (i = 0; i < rtcp_report->reception_report_count; i++) {
|
|
ast_free(rtcp_report->report_block[i]);
|
|
}
|
|
}
|
|
|
|
struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks)
|
|
{
|
|
struct ast_rtp_rtcp_report *rtcp_report;
|
|
|
|
/* Size of object is sizeof the report + the number of report_blocks * sizeof pointer */
|
|
rtcp_report = ao2_alloc((sizeof(*rtcp_report) + report_blocks * sizeof(struct ast_rtp_rtcp_report_block *)),
|
|
rtp_rtcp_report_dtor);
|
|
|
|
return rtcp_report;
|
|
}
|
|
|
|
void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
|
|
struct stasis_message_type *message_type,
|
|
struct ast_rtp_rtcp_report *report,
|
|
struct ast_json *blob)
|
|
{
|
|
RAII_VAR(struct rtcp_message_payload *, payload,
|
|
ao2_alloc(sizeof(*payload), rtcp_message_payload_dtor), ao2_cleanup);
|
|
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
|
|
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
|
|
|
|
if (!payload || !report) {
|
|
return;
|
|
}
|
|
|
|
if (!ast_strlen_zero(rtp->channel_uniqueid)) {
|
|
snapshot = ast_channel_snapshot_get_latest(rtp->channel_uniqueid);
|
|
if (snapshot) {
|
|
ao2_ref(snapshot, +1);
|
|
}
|
|
}
|
|
|
|
if (blob) {
|
|
ast_json_ref(blob);
|
|
}
|
|
ao2_ref(report, 1);
|
|
payload->snapshot = snapshot;
|
|
payload->blob = blob;
|
|
payload->report = report;
|
|
|
|
message = stasis_message_create(message_type, payload);
|
|
if (!message) {
|
|
return;
|
|
}
|
|
|
|
stasis_publish(ast_rtp_topic(), message);
|
|
}
|
|
|
|
/*!
|
|
* @{ \brief Define RTCP/RTP message types.
|
|
*/
|
|
STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_sent_type,
|
|
.to_ami = rtcp_report_to_ami,
|
|
.to_json = rtcp_report_to_json,);
|
|
STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_received_type,
|
|
.to_ami = rtcp_report_to_ami,
|
|
.to_json = rtcp_report_to_json,);
|
|
/*! @} */
|
|
|
|
struct stasis_topic *ast_rtp_topic(void)
|
|
{
|
|
return rtp_topic;
|
|
}
|
|
|
|
static void rtp_engine_shutdown(void)
|
|
{
|
|
ao2_cleanup(rtp_topic);
|
|
rtp_topic = NULL;
|
|
STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_received_type);
|
|
STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_sent_type);
|
|
}
|
|
|
|
int ast_rtp_engine_init()
|
|
{
|
|
struct ast_format tmpfmt;
|
|
|
|
ast_rwlock_init(&mime_types_lock);
|
|
ast_rwlock_init(&static_RTP_PT_lock);
|
|
|
|
rtp_topic = stasis_topic_create("rtp_topic");
|
|
if (!rtp_topic) {
|
|
return -1;
|
|
}
|
|
STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_sent_type);
|
|
STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_received_type);
|
|
ast_register_atexit(rtp_engine_shutdown);
|
|
|
|
/* Define all the RTP mime types available */
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0, "audio", "speex", 16000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0, "audio", "speex", 32000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
|
|
/* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
|
|
set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
|
|
set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
|
|
set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
|
|
/* Opus and VP8 */
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0, "audio", "opus", 48000);
|
|
set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0, "video", "VP8", 90000);
|
|
|
|
/* Define the static rtp payload mappings */
|
|
add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
|
|
#ifdef USE_DEPRECATED_G726
|
|
add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
|
|
#endif
|
|
add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
|
|
add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
|
|
add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
|
|
add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
|
|
add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
|
|
add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
|
|
add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
|
|
add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
|
|
add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
|
|
add_static_payload(13, NULL, AST_RTP_CN);
|
|
add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
|
|
add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
|
|
add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
|
|
add_static_payload(19, NULL, AST_RTP_CN); /* Also used for CN */
|
|
add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
|
|
add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
|
|
add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
|
|
add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
|
|
add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
|
|
add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
|
|
add_static_payload(101, NULL, AST_RTP_DTMF);
|
|
add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
|
|
add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
|
|
add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
|
|
add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0); /* Real time text chat (with redundancy encoding) */
|
|
add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0); /* Real time text chat */
|
|
add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
|
|
add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
|
|
add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
|
|
add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
|
|
add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
|
|
add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
|
|
add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
|
|
add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
|
|
add_static_payload(121, NULL, AST_RTP_CISCO_DTMF); /* Must be type 121 */
|
|
/* Opus and VP8 */
|
|
add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
|
|
add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
|
|
|
|
return 0;
|
|
}
|